Summoned Voices acts as a living memory of people and place. It consists of a series of door installations each with an intercom, sound system and a computer that is networked to a central file and database server. The design metaphor of the door presents a familiar scenario, that of announcing oneself at a doorway and waiting for a response from persons unknown. Signage instructs the public to speak, make sounds or sing into the intercom. Their voice is stored and interpreted, and results in local playback composed of the individual's voice with those that have gone before. Summoned Voices acts as an interpreter of sound, a message board and an imprint of a community - a place for expression, reflection and surprise.

Summoned Voices was premiered at the Art In Output Festival in Eindhoven, Netherlands in February 2003 and is a collaborative project by Iain Mott and Marc Raszewski. It was initiated during Iain Mott's artist residency at the CSIRO Mathematical and Information Sciences in 1999/2000 working with the Digital Media Information Systems (DMIS) research group in Sydney and Canberra. The project was assisted by the New Media Arts Fund of the Australia Council, the Federal Government's arts funding and advisory body. The Studium Generale of the Technische Universiteit Eindhoven assisted the final realisation at Art In Output.

Use the three scripts contained in the zip file below in "Download attachments" to batch convert a directory of B-format audio to binaural and UHJ stereo. Requires that SuperCollider is installed with the standard plugins and that the "Ctk" quark is enabled. The main shell script also encodes mp3 versions of the binaural and UHJ files and if this feature is used (not commented out), the system will require that "lame" is installed. The SuperCollider script uses the ATK and is taken directly from the SynthDef and NRT examples for ATK.

Unzip the scripts in a directory, edit the paths to match your installation and distribution of Linux in the file renderbinauralUHJ.sh, make this file executable and run in the directory containing the b-format files to perform the conversions.

The following procedure shows how to make B-format impulse responses (IRs) with the Linux software Aliki by Fons Adriaensen. A detailed user manual is available for Aliki, however the guide presented here in escuta.org is intended to show how to produce IRs without the need to run the software in the field and enables the use of portable audio recorders recorders such as the Tascam DR-680. The procedure was arrived upon through email correspondence with Fons. His utility "bform2ald" is included here with permission.

1. Launch Aliki in the directory in which you wish to create and store your "session" files and sub-directories, select the "Sweep" window and create a sweep file with these or other values:

rate: 48000

fade in: 0.1

start freq: 20

Sweep time: 10

End freq: 20e3

Fade out: 0.03

2. Select "Load" to load the sweep into Aliki and perform an export as a 24bit wav file or file type of your choosing.

3. Import the "*-F.wav" export in Ardour or other sound editor and insert an 800Hz blip or other audio marker 5 seconds before start. Insert some silence before the blip as some players (the Zoom H4n for example) may miss some initial milliseconds of files on playback. Export file as stereo 24bit 48kHz stereo file since the Zoom doesn't accept mono files.

4. Import file into Zoom H4n recorder for playback.

5. In the field, connect line out of Zoom H4n to Yorkville YSM1p and play file, recording with tetramic and Tascam DR-680. In my first test I recorded with the meter reading at around -16dB. Could have given the amp more gain, but the speaker casing was beginning to buzz with the low frequencies.

6. The Tascam creates 4 mono files. Use script to convert to A-format and with Tetrafile to convert to B-format with the mic's calibration data (with "def" setting).

8. Load the "ald" sweep capture into Aliki. Enter into edit mode and right-click to place a marker at the beginning of the blip. Use the logarithmic display to make the positioning easier. Once positioned, left-click "Time ref" to zero the location of the blip, then slide the marker to the 5 second mark and again left-click "Time ref" to zero the location of the start of the capture.

9. Right-click a second time a little to the right of the blue start marker. This will create a second olive coloured marker, marking the point at which a raised cosine fade-in starting at the blue marker will reach unity gain. When positioned, left-click "Trim start". Zoom out and drag the two markers to the end of the capture in order to perform a fade out in the same way with "Trim end". Use the log view to aid with this process.

10. Save this trimmed capture in the edited directory with "Save section".

11. Select "Cancel" and then "Load" to reload the freshly trimmed capture in the edited directory, then select "Convol". In this window, select the original sweep file used to create the capture in the "Sweep" dialogue. Enter "0" in the "Start time" field and in the "End time" field enter a number in seconds that represents the expected reverberation time plus two or three more seconds. Finally, select apply to perform the deconvolution, then perform a "Save section" to save the complete IR in the "impresp" directory.

12. Select "Cancel" and "Load" to load the recently created impulse in the "impresp" directory, then enter edit mode. The impulse may not be visible so use the zoom tools and in Log view, identify the first peak in the IR which should appear shortly after 0 seconds. This peak should represent the direct sound. While we may decide not to keep this peak, we will use it now to normalise the IR so that a 0 dB post fader aux send to the convolver will reproduce the correct ratio of direct sound to reverberation when using "tail IRs" or IRs without the direct impulse (see 13 below). To normalise, right-click to position the blue marker on the peak then left-click "Time-ref" to zero the very start of the direct impulse and shift-click "Gain / Norm".

13. The complete IR created above in step 12, containing the impulse of the direct signal as well as those of the first reflections and of the diffuse tail, may be convolved with an anechoic source to position that source in the sound field. If used in this way, the "dry" signal of the source should not be mixed with the "wet" or convolved signal and there will be no control over the degree of reverberation. If however the first 10msec of the IR are silenced (using the blue and olive markers and "Trim start" in Aliki to fade in from silence just before 10msec, for example), the anechoic signal may be positioned in the sound field by including the dry signal in the mix (panned by abisonic means to a position corresponding to that of the original source in the IR) and varying the gain on the "wet" or convolved signal to adjust the level or reverberation and reinforce the apparent position of the virtual source through first reflections encoded in the IR. Another alternative is to silence the first 120msec of the IR to create a so-called "tail IR". This removes the 1st reflections information entirely from the IR and enables the sound to be moved freely by ambisonic panning. The level of reverberation is adjustable however the will be no 1st reflections information to aid in the listener's localisation of the virtual source or to contribute to the illusion of its "naturalness". A fourth possibility is to use a tail IR in conjunction with various IRs for different locations. These IRs encoding first reflections only, those occurring between 10 and 120msec, could be chosen for example to match the positions of specific musicians on a stage. The engineer will first pan the dry signal of a source in a particular position, then mix in the wet signal derived from convolution with the 1st reflections IR for the corresponding location and additionally send a feed from the dry signal to a global tail IR common to all sources.

The script and other configurations detailed on this page convert mono files generated by a Tascam DR-680 with a Core Sound Tetramic soundfield microphone to B-format 4-channel wav files. It requires that the Tascam DR-680 is configured to save recordings as mono sound files on channels 1, 2, 3 & 4 and that these channel numbers match the corresponding capsules on the Tetramic. The script also requires that Tetramic calibration files are installed (see below) and the additional installation of the following programs by Fons Adriaensen:

Fons Adriaensen provides a free calibration service for Tetramics which generates calibration files specific to each microphone based on data provided with the microphone on purchase. See"TetraProc / TetraCal" and "Calibration service for Core Sound's TetraMic" on this page for further information.

Run the script in a directory containing the mono files. Change paths and configuration filenames in the script as necessary. Use the command line argument --elf to enable extended low frequency response in the b-format output (-3dB at 40Hz) or none to use the default roll-off at 80Hz.

The B-format script is contained in the attachment "mono2bformat.zip" below. Alternatively, copy the following code:

#!/bin/bash

#Converts dated mono files generated by a Tascam DR-680 with a Coresound Tetramic ambisonic microphone to B-format 4-channel wav files. Run this script in directory containing the mono files. Change paths as necessary. Use the command line argument --elf to enable extended low frequency response in the b-format output (-3dB at 40Hz) or none to use the default roll-off at 80Hz.

Mosca is a SuperCollider class for GUI-assisted authoring of ambisonic sound fields with simulated moving or stationary sound sources. The class makes extensive use of the Ambisonic Toolkit (ATK, see: http://www.ambisonictoolkit.net/) by Joseph Anderson and the Automation quark (https://github.com/neeels/Automation) by Neels Hofmeyr. Mosca is written by Iain Mott and licensed under a Creative Commons Attribution-NonCommercial 4.0 International License: http://creativecommons.org/licenses/by-nc/4.0/

Sound fields may be decoded using a variety of built in 1st order ambisonic SuperCollider decoders (including binaural) or with external 2nd order decoders such as Ambdec in Linux. Input sources may be any combination of mono, stereo or B-format material and the signals may originate from file, from hardware inputs (physical or from other applications such a DAW via Jack) or from SuperCollider's own synths. In the case of synth input, synths are associated by the user with a particular source in the GUI and registered in a synth registry. In this way, they are spatialised by the GUI and also receive all of the data from the GUI pertaining to the source (eg. x, y and z coordinates and auxiliary fader data). Mosca has its own transport provided by the Automation quark for recording and playback of source data. This may be used independently or may be synchronised to a DAW using Midi Machine Control (MMC) messages. This function has been tested to work with Ardour and Jack.

Mono and stereo sources are encoded as second order ambisonic signals whereas B-format signals remain as 1st order and are angled in space using "push" transformations. Source signals are attenuated proportionally to the inverse of the square root of proximity or in a linear relationship with distance, selectable on a per-source basis via the GUI. All sources are subject to high-frequency attenuation with distance and if decoding is performed by one of the ATK's 1st order decoders, a proximity effect is generated adding a bass boost to proximal sources among other phase effects to simulate wave curvature (see: http://doc.sccode.org/Classes/FoaProximity.html).

Reverberation is performed either using a B-format tail room impulse response (RIR) - the preferred method - or using simple built-in allpass filters, options selectable on creation of a Mosca instance. With both options, two reverberation level controls are included in the GUI to set close and distant levels. A further two reverb types are selectable in the GUI on a per-source basis for both RIR and allpass reverberation modes. The default reverb type uses John Chowning's technique of applying "local" and "global" reverberation to sources (CHOWNING). The "Close" reverberation of the GUI in this case is "global" and is audible by the listener from all directions when the source is close whereas "distant" reverb is "local" in scope and is encoded as a 2nd order ambisonic signal along with the dry signal. This predominates as the source becomes more distant. The second type of reverberation may be described as a "2nd order diffuse A-format reverberation". This technique produces reverberation weighted in the direction of sound events encoded in the dry ambisonic signal and involves conversion to and from A-format in order to apply the effect (ANDERSON). The encoded 2nd order ambisonic signal is converted to a 12-channel A-format signal and then either a) convolved with a B-format RIR which has been "upsampled" to 2nd order and converted to A-format impulse spectrum, or, as in the case of the allpass option, b) passed through a 12-channel bank of allpass filters before being converted back to a 2nd order B-format diffuse signal. Please note that the 2nd order diffuse reverberation may require the user to set a larger audio output buffer and thus increase the latency of the system. The "Chowning" type reverberation is more efficient and the "allpass" option, more still.

Mosca also has other features including a scalable Doppler Effect on moving sources, looping of sources loaded from file, adjustment of virtual loudspeaker angle of stereo sources and in the case of B-format sources: a rotation control, adjustment of "directivity" (see ATK documentation) and a control of "contraction", whereby the B-format signal may be crossfaded with its W component and which is spatialised as a 2nd order ambisonic signal.

Additionally, Mosca supports methods for making "A-format inserts" on any source spatialised in the GUI. In this way, the user may write a filtering synth and apply it to the sound without disrupting the encoded spatial characteristics.

If you use these resources or have suggestions, please get in contact!

You may clone the Mosca quark using "git clone https://github.com/escuta/mosca" or download the project as a Zip file from the github page and then place it in you quarks directory to install. Alternatively, if using Supercollider 3.7 or higher, simply run the following command in SuperCollider to install: Quarks.install("https://github.com/escuta/mosca");

See the Mosca quark help file for instructions. You may also choose to download the following zipped example project directory:

This archive contains the file structure necessary to run Mosca as well as example room impulse responses (RIRs). B-format material, including a B-format Spitfire recording by John Leonard, is also provided in the archive with kind permission as well as other B-format material recorded by Iain Mott in Chapada dos Veadeiros and Brasilia. A README file in the archive and attached separately below, includes complete instructions on installing Mosca from scratch on Linux. Importantly, the README also details how to use the Mosca GUI.

Squeezebox incorporates spatial sound, computer graphics and kinetic sculpture. Participants manipulate the sculpture to produce real-time changes to the spatial location and timbre of the sound, as well as to manipulate digitised images. The sound and images are presented as an integrated plastic object, a form which is squeezed and moulded by participants. The artwork consists of a frame supporting four sculpted pistons on pneumatic shafts. An interactive image is displayed on a monitor beneath a one-way mirror at the centre of the sculpture. Four loudspeakers are situated at the outer four corners.The cast hands of Squeezebox invite participation. Participants grasp and press down the sculpted pieces, working against a pneumatic back-pressure to elicit both sound and image. The interaction reveals a form which has visual, aural as well as physical properties. As participants press down on the hands a sound mass is shifted from one point of the sculpture to another by pressing down on alternate pistons. Music is produced algorithmically and is derived from a set of rules which respond to the spatial location of the sound mass. The system of rules however is never static. One spatial strategy gives way to another resulting in an evolution of sound, requiring a constant readjustment of focus in the listener.

Squeezebox is collabroration between Iain Mott, Marc Raszewski and artist Tim Barrass who designed the interactive graphics. It was first exhibited in "Earwitness", Experimenta '94, ether ohnetitel, Melbourne, 1994. The project was produced with the assistance of The Australia Council, the Federal Government's arts funding and advisory body.

The composition Pope's Eye by Iain Mott was an outcome of a two week artist residency undertaken by Ros Bandt and Iain Mott in 2004 at the Melbourne Aquarium. Bandt and Mott made hydrophone recordings at the aquarium as well as recordings at Pope's Eye in Port Phillip Bay, Victoria. The aquarium recordings were subject to high levels of noise from the filtration equipment and noise reduction software was used to isolate the marine sounds. The sounds that can be heard in this composition include feeding sounds of marine life (fish and crustaceans), the sounds of fish calls, the sounds of staff divers at the aquarium, a motor boat on the bay and gannets at Pope's Eye. Other than the noise reduction, very little audio processing was applied to the recorded sound. The sounds were simply edited into a narrative form.

The Great Call is an early composition by Iain Mott made at La Trobe University. It was originally composed for the University of Melbourne Guild Dance Theatre's production of Signals in 1989. It was later performed as part of the Astra concert program for 1990-10-05 at Elm St Hall, North Melbourne. The composition is based on recordings of the homonymous vocalisations of the white cheeked gibbon (Hylobates concolor) made at the Melbourne Zoological Gardens in March 1998 on analogue tape. In the studio a pitch to MIDI tracking device was used to control an Oberheim Xpander synthesiser and a sampler. Other synthesised sounds, percussion and vocalisations were improvised and recorded by the composer on multitrack tape.